Issues 15000 - 15999

[..]
ASTERISK-15000: Phone init problem solved with backport.
ASTERISK-15001: chan_sip breaks RFC by incrementing session version between non-reliable 1xx and 200
ASTERISK-15002: Crash on many connected / canceled calls
ASTERISK-15003: [patch] Add couple of useful extensions as examples
ASTERISK-15004: [patch] Security Problem
ASTERISK-15005: Crash in chan_dahdi with ss7 support
ASTERISK-15006: SIP subscriptions are lost after a reload
ASTERISK-15007: [patch] iax2 show cache, locks channels.
ASTERISK-15008: With delayreject=yes calls sometimes fail
ASTERISK-15009: Exceptionally long voice queue length queuing to Local/XXXXXXX
ASTERISK-15010: [patch] asterisk crashes when there are no RTP port left
ASTERISK-15011: Can't dial to exchange UM 2007
ASTERISK-15012: [regression] 1.4 SVN branch does not write Userfield in CDR
ASTERISK-15013: [patch] default say.conf for new number method doesnt handle all numbers
ASTERISK-15014: Sometimes macro in h extension returns to s extension
ASTERISK-15015: app_voicemail appending IMAPFOLDER to 'vm-' to create filename for prompt to play.
ASTERISK-15016: [patch] incorrect playback when using say_date_with_format_es on one o'clock (spanish)
ASTERISK-15017: [patch] Hangup extension executed twice in 1.6.2 RC2
ASTERISK-15018: Kapanga softphone exposes bug in SIP channel driver
ASTERISK-15019: Application Extenspy
ASTERISK-15020: Cant delete temporary greetings
ASTERISK-15021: [patch] Auto-fallthrough when attempting to enter DTMF using Background() in a Macro()
ASTERISK-15022: Lockup in chan_sip
ASTERISK-15023: [patch] Fix/improve transaction/dialog-matching in pedantic mode
ASTERISK-15024: G726 is choppy on IAX - 1.6
ASTERISK-15025: ExternalIVR without argument causes segmentation fault
ASTERISK-15026: SIP peers are not being built from users.conf configuration
ASTERISK-15027: [patch] ExternalIVR TCP client functionality does not work
ASTERISK-15028: RTP Media Port Change Ignored
ASTERISK-15029: [patch] ExternalIVR trapping non-existent files does not work
ASTERISK-15030: Trying to send a fax with zoiper (t38) causes "buffer overflow message too long"
ASTERISK-15031: new libpri features not detected with a custom libpri location
ASTERISK-15032: hylafax(+iaxmodem)+ReceiveFAX leads to "Audio loop reports T38 switchover but t38state != T38_STATE_NEGOTIATED"
ASTERISK-15033: overlap dial from BRI phone: unlimited number of digits
ASTERISK-15034: flags not initalized in app_softhangup, causes undefined behavoir
ASTERISK-15035: Asterisk crashes randomly with a segmentation fault in __res_vinit ()
ASTERISK-15036: [patch] Service indicator not managed
ASTERISK-15037: ChanSpy crashes Asterisk
ASTERISK-15038: spaces in caller id name cause unexpected behavior
ASTERISK-15039: [patch] ast_gethostbyname doesn't set h_length if argument is an IP Address
ASTERISK-15040: Meetme - Quitting time issue
ASTERISK-15041: [patch] ExternalIVR Does not use copy IP Address correctly
ASTERISK-15042: [patch] realtime function does not return pair when database value is null
ASTERISK-15043: CVE-2008-7220: static-http/prototype.js is vulnerable to "cross-site ajax requests"
ASTERISK-15044: hint not updated correctly on outgoing SIP calls
ASTERISK-15045: IAX trunk clicks as other calls in the same trunk hang up
ASTERISK-15046: Literal values wrapped in documentation
ASTERISK-15047: Need a CLI command to force a reconnect of ODBC connections
ASTERISK-15048: [patch] ExternalIVR does not return event for file when file playing is interrupted
ASTERISK-15049: Callerid is not recorded in database
ASTERISK-15050: MOH silence
ASTERISK-15051: Unable to create/find SIP channel for this INVITE
ASTERISK-15052: [patch] MixMonitor thread doesn't exit until channel is dropped
ASTERISK-15053: [patch] Extend slin16 support to SIP calls
ASTERISK-15054: hangup on transfer
ASTERISK-15055: [patch] Use pkg-config to find gmime libraries.
ASTERISK-15056: Asterisk core-dumps if used as loadgenerator using callfiles
ASTERISK-15057: [patch] Wrong cause value for 'answered elsewhere'
ASTERISK-15058: [regression] High CPU usage, choppy sound
ASTERISK-15059: Calltoken and Realtime
ASTERISK-15060: Crash on meetme leave
ASTERISK-15061: [patch] incorrect 'core show channel channel-name' output
ASTERISK-15062: asterisk continiously crashes when iax-call received
ASTERISK-15063: [patch] Limit on simultaneous incoming calls for queue members
ASTERISK-15064: [patch] Setting dialplan hint and using a global variable gives incorrect warning.
ASTERISK-15065: Chaspy always active Option ´o´
ASTERISK-15066: full system crash every other day
ASTERISK-15067: [patch] temporary greetings can't be erased in 1.4
ASTERISK-15068: AgentComplete event sent as soon as call is answered in call to queue through local channel
ASTERISK-15069: [patch] "make config" creates really wrong runlevels in Debian (includes patch)
ASTERISK-15070: [patch] chan_mgcp new feature: digitmaps definitions
ASTERISK-15071: [patch] Asterisk does not fully support SIP connections to Internet Telephony Service Providers
ASTERISK-15072: Revision 202007 Introduces Deadlock
ASTERISK-15073: OpenSolaris Build Problem with editline
ASTERISK-15074: Crash in SQLAllocHandle
ASTERISK-15075: res_pktccops.c using MSG_NOSIGNAL
ASTERISK-15076: Asterisk Crash after SIP Transfer
ASTERISK-15077: [regression] Whisper mode in ChanSpy() has delays and gaps in audio (sometimes not working at all)
ASTERISK-15078: Dial Option D() does not execute in parallel witb option A()
ASTERISK-15079: [patch] Thousands of Invites never discarded in sip channels
ASTERISK-15080: [patch] out of dialog SIP_NOTIFY with event='keep-alive'
ASTERISK-15081: Inband onhook ringing not applying indications.conf settings
ASTERISK-15082: Crash of Outgoing Call
ASTERISK-15083: [patch] T.38 reinvite fails after receiving "415 Unsupported media type" when it could continue in audio mode
ASTERISK-15084: [patch] RFC 4474 Implementation of SIP identity
ASTERISK-15085: chan_mobile pairs, dials, and receives calls, but no audio
ASTERISK-15086: [patch] if, for, while, switch statements all missing space, - Coding guidelines
ASTERISK-15087: chan_mobile.c needs updating for 32->64 bit changes
ASTERISK-15088: [patch] Segfault with limit data L(x:y) and verbosity >= 3
ASTERISK-15089: IAX2 Codec negotiation fails since 227580
ASTERISK-15090: Can't compile H323 channel driver
ASTERISK-15091: Core dump in audio_audiohook_write_list
ASTERISK-15092: Core dump in vsnprintf / ast_rtp_get_quality / sip_hangup
ASTERISK-15093: WARNING channel.c __ast_channel_alloc_ap: Channel allocation failed: Can't create alert pipe!
ASTERISK-15094: Outbound proxy with realtime integration not working
ASTERISK-15095: sound files missing for say_enumeration: digits/h-hundred
ASTERISK-15096: T.38 Fax Termination Failing
ASTERISK-15097: [patch] Asterisk 1.6.0.13 Asterisk crashes when running dialplan app macro on a macro that does not exist.
ASTERISK-15098: [patch] Backport patch 9048 to v1.4: Provide colors for daemonized asterisk
ASTERISK-15099: Call failed to go through, reason (8) Congestion (circuits busy) Description: When launching more t
ASTERISK-15100: Asterisk Freezes when more than 5 simultaneous calls using chan_mobile
ASTERISK-15101: [patch] Segfault in chan_iax2.so when receiving call without CallToken support
ASTERISK-15102: [patch] asterisk keeps starting new processes for streaming audio MOH
ASTERISK-15103: Call Hold not working on some phones
ASTERISK-15104: [patch] When using ooh323h it is impossible to call when number have more than 3 number...
ASTERISK-15105: Setting a call-forward on an analog phone results in the analog phone still being rung when dialed
ASTERISK-15106: [patch] chan_ooh323 don't works with Avaya Definity
ASTERISK-15107: [patch] Event collision in ExternalIVR resolved by documenting issue
ASTERISK-15108: [regression] Early audio message doesn't play over SIP
ASTERISK-15109: [patch] Script to automatically email backtrace to admin
ASTERISK-15110: A func_math WARNING for a operation that is not executed in ExecIf
ASTERISK-15111: [patch] Invalid behaviour of Return within Gosub and AGI
ASTERISK-15112: [patch] #define MAX_LANGUAGE increment from 20 to 30 in include/asterisk/channel.h
ASTERISK-15113: Asterisk does not connect the call to internal extension
ASTERISK-15114: Crash revision 229091 in audiohook_inheritance_destroy
ASTERISK-15115: [patch] Fix ExternalIVR Documentation in 1.4
ASTERISK-15116: T.38 passthrough issue in 1.6.1.6
ASTERISK-15117: Undefined references in function `agent_set_base_channel':
ASTERISK-15118: [patch] CDR always set disposition as NO ANSWER with .call files
ASTERISK-15119: [patch] "requirecalltoken" config directive not respected globally
ASTERISK-15120: [patch] Remove features from ExternalIVR documentation
ASTERISK-15121: 1.4.26.3 security issue - Chinese IPs somehow are making calls without authentication
ASTERISK-15122: Call files produces "NO ANSWER" record
ASTERISK-15123: eswitch does not substiotute variables when using Local Channel
ASTERISK-15124: ERROR[24164]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
ASTERISK-15125: [patch] g726 to g726aal2 translation and cost-calculation are wrong but easily fixed.
ASTERISK-15126: Failed assertion in chan_iax2 update_registry/ast_sched_del
ASTERISK-15127: [patch] Call parking via AMI causes announcment and ringback to caller channel
ASTERISK-15128: Placing a URI call fails when URI string contains a non-standard port
ASTERISK-15129: Can't initiate outbound SIP calls when siren14 is the ONLY enabled codec
ASTERISK-15130: [patch] Building Queues with Asterisk - A How-to Guide
ASTERISK-15131: sip calls drop because of BYE's
ASTERISK-15132: AMD() incorrectly sets AMDCAUSE channel variable
ASTERISK-15133: [patch] QUEUE_MEMBER(...,free) counts wrapping-up agents as available
ASTERISK-15134: [patch] issues in processing "Action: Events" eventmask
ASTERISK-15135: php agi causes segfaults in php -> /var/lib/asterisk/agi-bin/fixlocalprefix
ASTERISK-15136: [patch] Iconv is a glibc-ism, and as such should be called out explicitly
ASTERISK-15137: [patch] [regression] The status of External SIP peer used as Queue member is not updating correctly
ASTERISK-15138: ast_rtp_destroy causes segmentation violation
ASTERISK-15139: [patch] Muted user remains talking forever
ASTERISK-15140: detect fsk caller-id with DT-AS start signal
ASTERISK-15141: UDPTL asked to send 77 bytes of IFP when far end only prepared to accept 45 bytes; data loss may occur.
ASTERISK-15142: Symbol referencing errors (MIN/MAX in channel.o/udptl.o)
ASTERISK-15143: ODBC Crash in 1.6.0.18-rc2
ASTERISK-15144: hello
ASTERISK-15145: SIP REFER initiated from the Asterisk Transfer application fails on a SOPHO iS3000 SIP Server
ASTERISK-15146: DAHDIScan() only returns dead air
ASTERISK-15147: MixMonitor stops audio on native bridging
ASTERISK-15148: [patch] Memory leak in res_config_ldap when using realtime extensions
ASTERISK-15149: Missing CDR
ASTERISK-15150: core dump somewhere in format_wav or res_musiconhold (1.4.27-RC2)
ASTERISK-15151: [patch] Allow execincludes within asterisk.conf
ASTERISK-15152: [patch] Conditional jump or move depends on uninitialised STACK value
ASTERISK-15153: ChanSpy Whisper not working properly when peer has VAD and CNG on.
ASTERISK-15154: [patch] VM_DATE does not follow emaildateformat format for pager email
ASTERISK-15155: [patch] UserEvent manager action is not ACKed
ASTERISK-15156: Etisalat UAE Disconnect Tone
ASTERISK-15157: [patch] Trunk won't build as cross-compilation
ASTERISK-15158: [patch] unanswered has no effect
ASTERISK-15159: [patch] Last line of SDP is not being parsed
ASTERISK-15160: incomming call on agent when an agent is in outgoing call.
ASTERISK-15161: [patch] Asterisk doesn't free udp ports
ASTERISK-15162: [patch] message limit (maxmsg) can be exceeded in 1.6.x creating orphan voicemail
ASTERISK-15163: [patch] Language code collisions for certan languages
ASTERISK-15164: crash and core dump
ASTERISK-15165: Asterisk 1.6.0.13 Asterisk crashes intermittently cause unknown.
ASTERISK-15166: [patch] response to "Action: Events" is not finished by empty line
ASTERISK-15167: No way to set pin for new ConfBridge conferences.
ASTERISK-15168: Outoing calls disconnected immediately after remote end picks up.
ASTERISK-15169: [patch] Incoming multiline SMS causes chan_mobile to stop working
ASTERISK-15170: [patch] asterisk reload causes mpg123 streams to be recreated
ASTERISK-15171: RTPAUDIOQOS and RTPAUDIOQOSBRIDGED false statistics
ASTERISK-15172: Unable to negotiate codecs using IAX2 (and possibly others)
ASTERISK-15173: Double CDR if unanswered call
ASTERISK-15174: somtimes when agent is at conversation with queue caller, call disconnected and new person begin his conversation
ASTERISK-15175: [patch] Support for disabling automon selectively per peer
ASTERISK-15176: Review of internal_timing code
ASTERISK-15177: Implement a pin auth for ConfBridge, like conferences in meetme.conf have
ASTERISK-15178: G723 codec has digitzed voice
ASTERISK-15179: [patch] core show function CDR reports wrong disposition values
ASTERISK-15180: ast_ouraddrfor doesn't do htons() on the port
ASTERISK-15181: app_voicemail.c strip_control() strips more than just control chars
ASTERISK-15182: [patch] [regression] Asterisk sip.conf realtime register, contact problem
ASTERISK-15183: [patch][regression] DTMF Not Recognized with Exchange UM
ASTERISK-15184: [patch] G.719 Pass-through Support for Asterisk
ASTERISK-15185: Outgoing Caller ID Name For QSIG
ASTERISK-15186: [patch] handle_incoming() incorrectly sets p->method to SIP_ACK
ASTERISK-15187: [patch] menuselect.makeopts: does not properly unselect an option with a leading - (minus)
ASTERISK-15188: [patch] Timeout in SPEECH RECOGNIZE not working.
ASTERISK-15189: [patch] After upgrading to asterisk 1.4.27 Optipoint SIP phone can no longer register
ASTERISK-15190: [patch] pedantic sip checking needed to generate valid messages (but broken)
ASTERISK-15191: 64bit Host OS, 32bit OpenVZ/Virtuozzo VPS, Dahdi does not work
ASTERISK-15192: Asterisk Reference Information has incomplete coverage of sip.conf file
ASTERISK-15193: [patch] Asterisk crashes with Asyc AGI when hangup
ASTERISK-15194: [patch] Some warnings when parsing extensions.conf fail to include line numbers
ASTERISK-15195: g729 doesnt work when asterisk installed with ([*] LOW_MEMORY)
ASTERISK-15196: [patch] ExternalIVR confuses AGI by double closing FDs
ASTERISK-15197: Interlock in chan_dahdi
ASTERISK-15198: IAX calls drop after ~30 seconds between 1.4.27rc5 and 1.2.36
ASTERISK-15199: [regression] Old/new message Seen/Unseen is not RFC compliant
ASTERISK-15200: segfault in "core show functions"
ASTERISK-15201: 488 not acceptable when receiving T.38 at 14400 speed
ASTERISK-15202: Recursion crash in pbx_ael.c
ASTERISK-15203: dummy_start (data=0xb7c9be00) at utils.c and in clone () from /lib/libc.so.6
ASTERISK-15204: Responds sendrecv to recvonly SDP, but RFC 3264 says sendonly and inactive are only possible replies
ASTERISK-15205: When placing an external SIP trunk caller on hold, no music on hold audio is heard
ASTERISK-15206: Useless MySQL queries when doing sip qualify
ASTERISK-15207: [regression] After upgrading i see a lot more Notices peers status
ASTERISK-15208: [patch] stop the flame - remove 'silly' from channel.c
ASTERISK-15209: [patch] New SDP handling code totally broke T.38 reinvites
ASTERISK-15210: SRV registration
ASTERISK-15211: [patch] When using 'joinempty=strict', "in use" devices not seen as "unavailable".
ASTERISK-15212: [patch] Incorrect reloading of realtime peer causes mailbox list to expand indefinitely
ASTERISK-15213: [patch] [regression] asterisk deadlocks and calls will stop queueing.
ASTERISK-15214: segfault if too many rooms in meetme.conf
ASTERISK-15215: [patch] confusing description in configs/sip.conf.sample
ASTERISK-15216: Autodestruct
ASTERISK-15217: Siemens S685IP g722 gets not translated
ASTERISK-15218: Asterisk responds 488 - Not acceptable here on T38 reinvite
ASTERISK-15219: Asterisk drops call leg to Cisco while other leg remains up.
ASTERISK-15220: chan_sip transforms %23 to # (UTF-8 issue) in Contact field
ASTERISK-15221: Asterisk is crashing on any H323 call.
ASTERISK-15222: [patch] A blind transfer results in a call with empty accountcode in a specific circumstance
ASTERISK-15223: When trying to enable jitter buffer on local channel atserisk crash
ASTERISK-15224: Interlock between directed pickup and device state threads
ASTERISK-15225: Segmentation fault in chan_sip in function initreqprep
ASTERISK-15226: [patch] Asterisk crashes randomly on mISDN RELEASE_COMPLETE
ASTERISK-15227: reinvites fail when sdp-session does not increment
ASTERISK-15228: [patch] Segmentation Fault on Originate command.
ASTERISK-15229: [patch] chan_mobile doesn't hangup
ASTERISK-15230: RFC 2833 DTMF Events Generated by Polycom IP Phone Running v3.2.1 F/W Not Recognized by Asterisk 1.4
ASTERISK-15231: [patch] Fix bootstrap.sh on OpenSolaris
ASTERISK-15232: [patch] configure fails to detect spandsp/expose.h when not in system include path
ASTERISK-15233: [patch] [branch] New CLI command: manager show settings
ASTERISK-15234: pbx_extension_helper cannot find labels in contexts
ASTERISK-15235: Call terminates 5 seconds after establishing
ASTERISK-15236: chan_dahdi uses 1-2 for second port on one span
ASTERISK-15237: Custom CDR values not logged when dialing with local channels
ASTERISK-15238: [patch] get_sdp_line condition is not right?
ASTERISK-15239: System completely hangs after executing an ODBC function
ASTERISK-15240: [patch] Deleting Multiple IMAP voicemails does not work reliably
ASTERISK-15241: Asterisk 1.6.1 won't "answer" the phone when using a callcentric sip trunk
ASTERISK-15242: transmit_refer leaks sip_refer structures
ASTERISK-15243: [patch] keepstats option removed when it shouldn't
ASTERISK-15244: Segfault in ast_frdup
ASTERISK-15245: [patch] Hints do not have the correct state on initialization
ASTERISK-15246: Interlock between SIP and device state threads
ASTERISK-15247: [patch] Send Manager Event on SNOM X-ClientCode SIP INFO message
ASTERISK-15248: [patch] new option: lockconfdir for protecting conf files in /etc/asterisk during reloads
ASTERISK-15249: [patch] Only the last setvar is effective for a given channel
ASTERISK-15250: LOCK behaves like trylock (not waiting for 3 seconds)
ASTERISK-15251: [patch] Asterisk crashes after receiving fax with 'double free'
ASTERISK-15252: The current SVN does not compile
ASTERISK-15253: [patch] Asterisk crash when SpeechCreate() is used in dialplan without exact name of the module, using the default.
ASTERISK-15254: [patch] Since changeset 231437: Queue ERROR[7429]: astobj2.c:114 INTERNAL_OBJ: bad magic number 0x0 for 0xb7174c50
ASTERISK-15255: SendFax sessions not correctly reported
ASTERISK-15256: Asterisk detects DTMF inband even when dtmfmode=rfc2833
ASTERISK-15257: Asterisk crash in rtp.c a few times a day
ASTERISK-15258: Asterisk ignoring sendonly SDP generated from Cisco UCM after generating inactive SDP when a Cisco phone initiates hold
ASTERISK-15259: Crash due to fault about twice daily
ASTERISK-15260: Crash when performing dial
ASTERISK-15261: [patch] Incorrect path passed to MONITOR_EXEC application after 'Monitor()' call finishes.
ASTERISK-15262: thereis no sample for asterisk.conf
ASTERISK-15263: res_monitor.c chan->monitor->filename_base has duplicated path
ASTERISK-15264: [patch] mpg123 <defunct>
ASTERISK-15265: [patch] Patch: New admin features: Roll call, eject all, mute all, record in-conf
ASTERISK-15266: [patch] Implicit declaration of 'ast_complete_source_filename' and 'ast_rtp_destroy' with LOW_MEMORY enabled in trunk
ASTERISK-15267: About issue-0012950 [patch] PacketCable NCS 1.0 Support for Docsis / Eurodocsis Networks..
ASTERISK-15268: [patch] Add support for ring indication when calling member
ASTERISK-15269: [patch] app_queue segfaults if realtime field uniqueid is NULL
ASTERISK-15270: [patch] Missing session level connection data (c=) breaks process_sdp()
ASTERISK-15271: [patch] New music on hold patches cause asterisk + full system hard lock
ASTERISK-15272: [patch] busydetect incorrectly hangs up incoming call due to incoming DTMF seen as busy pattern.
ASTERISK-15273: Asterisk 1.6.1.9 lockup when caller hangs up in StartMusicOnHold()
ASTERISK-15274: [patch] App MeeMe Set the channels' account code to the conference room number
ASTERISK-15275: vsendonly is write only
ASTERISK-15276: Build fails on OpenBSD4.2 in utils.o
ASTERISK-15277: [patch] Segfault in res_config_ldap
ASTERISK-15278: Error in ulaw
ASTERISK-15279: [patch] SIP Realtime SQL Table
ASTERISK-15280: [patch] rtpkeepalive parsed twice
ASTERISK-15281: [patch] cdr_mysql driver does not have an option to log in GMT time
ASTERISK-15282: dahdi show channels does not show an outgoing call
ASTERISK-15283: [patch] New option setvarout which sets channel variable for outbound channels to a peer
ASTERISK-15284: [patch] Missing \n in logging
ASTERISK-15285: segfault error 4 in libpthread on Ubuntu
ASTERISK-15286: [patch] potential buffer overflow in say_date_with_format()
ASTERISK-15287: Asterisk runs out of handles because of stuck SIP dialogs
ASTERISK-15288: [patch] First DTMF digit is missed if pressed during "using your touch tone keypad..." announcement
ASTERISK-15289: [patch] IP and port is not transferred for t.38
ASTERISK-15290: Ignoring unknown format wav & wav49...
ASTERISK-15291: T.38 Fax Termination Failing Resolution -- Request for More Information
ASTERISK-15292: [patch] Message forwarding with prepention does not backup original message and length as intended
ASTERISK-15293: [patch] Portability tweaks to contrib/scripts/safe_asterisk
ASTERISK-15294: ChanSpy and ExtenSpy applications don't accept a colon-delimited list of groups
ASTERISK-15295: one way audio after call waiting
ASTERISK-15296: Asterisk 1.6.2: CDR is not produced with .call files
ASTERISK-15297: Asterisk ignore SIP signalling path
ASTERISK-15298: SIP qualify fails unless NAT is enabled
ASTERISK-15299: [patch] remainder ast_expr2 func misspelt as reminder
ASTERISK-15300: [patch] park() function takes 100% of CPU
ASTERISK-15301: [patch] Callee on outside line can take parking and forwarding rights
ASTERISK-15302: Asterisk crashes on dtmf detection on channel with 2 bluetooth cellphone
ASTERISK-15303: [patch] Send manager event on Call Pickup
ASTERISK-15304: racecondition leading to deadlock in chan_local
ASTERISK-15305: app_voicemail will module will not load
ASTERISK-15306: [patch] Background() when called from AGI script no longer gives digit code when interrupted
ASTERISK-15307: Crash when making outbound call
ASTERISK-15308: asterisk is not able to register with SIP server
ASTERISK-15309: [patch] Unable to escape back to dialplan or operator, using 'o' and 'a' extensions in dialcontext
ASTERISK-15310: [regression] DTMF Tones not working
ASTERISK-15311: [patch] silencethreshold=0 when dsp.conf not existing
ASTERISK-15312: Can't handle frames in 2 format - revisited
ASTERISK-15313: chan sip removes peers like if srvlookup were active, but it is not
ASTERISK-15314: Asterisk wrongfully sends 403 instead of 401
ASTERISK-15315: segfault error 4
ASTERISK-15316: Voicemail messages flagged as urgent do not get emailed
ASTERISK-15317: Trunk does not compile (again) on Darwin (MacOS 10.5)
ASTERISK-15318: .call file does not create cdr
ASTERISK-15319: core show hints do not follow the general sorting ordre
ASTERISK-15320: [patch] Lots of crashes after upgrading to latest 1.6.0.20-rc1
ASTERISK-15321: the value of odbcstorage is not taken into account
ASTERISK-15322: [patch] ASTARGS in sysconfig not inherited as startup options
ASTERISK-15323: [patch] Manager hooks don't execute if there aren't any manager sessions
ASTERISK-15324: IAX2 Can't compress subclass 4294967295
ASTERISK-15325: [Patch] always m=text 0 in sdp answer
ASTERISK-15326: Variables not Passed
ASTERISK-15327: [patch] Change in sip show channels display format allowing more digits for CID
ASTERISK-15328: [patch] DTMF CallerID detection without polarity reversal
ASTERISK-15329: [patch] for reading and writing to text file
ASTERISK-15330: [regression] Record application hangs up after exactly 30 seconds, with or without silence or duration specified
ASTERISK-15331: make menuselect fails due to undefined symbols (initscr32, w32addch) in menuselect_curses.o
ASTERISK-15332: [patch] Set time for use in evaluating holiday dialplans
ASTERISK-15333: Asterisk sends port 0 in the Register message to Proxy if listen port is dynamic
ASTERISK-15334: [patch] ast_filecopy: keep modes for the created file
ASTERISK-15335: [patch] cidname and cidnum in output of "sip show peers"
ASTERISK-15336: [patch] missing newline after reply to event manager-action
ASTERISK-15337: [patch] [regression] Custom devsate set to INUSE but shows as UNAVAILABLE when accessing through the dialplan
ASTERISK-15338: "sip show peers" returns notice
ASTERISK-15339: [patch][regression] VMAuthenticate not playing greeting
ASTERISK-15340: [patch] Serious problem with pattern matching (regression in #15421)
ASTERISK-15341: ReceiveFAX always end with "Transmission error" but Fax transmitted successfully
ASTERISK-15342: Aastra phones won't register SIP
ASTERISK-15343: after udp error sip phones get kicked
ASTERISK-15344: [patch] Send manager event on Call Forward
ASTERISK-15345: [regression] Build fails when defs are required by the linker
ASTERISK-15346: [patch] When setting a soundfile for announce with a length longer then 80 chars a storage overlay happens
ASTERISK-15347: [patch] Missing plus signs in MAKE/SUBMAKE calls prevent parallel make from operating correctly
ASTERISK-15348: CLI reports wrong data
ASTERISK-15349: 1.6.1.12-rc1 crash around 100 SIP call setup with media
ASTERISK-15350: [patch] utils/extconf.c growing apart from main/pbx.c
ASTERISK-15351: one peer for SIP provider with SRV
ASTERISK-15352: [patch] [regression] T.38 no longer functions
ASTERISK-15353: [patch] Added musiconhold class in manager event
ASTERISK-15354: No hold event is generated for a second call on a SIP channel
ASTERISK-15355: [patch] port in users.conf is not honored in the register statement
ASTERISK-15356: [patch] Some small solaris fixes (threadstorage / moh)
ASTERISK-15357: Segfault in res_agi with no second paramter to EXEC
ASTERISK-15358: [patch] Asterisk man page outdated
ASTERISK-15359: [patch] Segmentation fault using manager http MXML
ASTERISK-15360: Exceptionally long voice queue length with chan_iax2 + res_timing_pthread causes high CPU usage
ASTERISK-15361: [regression] app_sms not working in 1.6.1.12 (same as 0012779)
ASTERISK-15362: [patch] meetme can support only 6341 rooms
ASTERISK-15363: [patch] Local values not set within gosub
ASTERISK-15364: [patch] Monitor resumes recording after SIP transfer despite StopMonitor() having been called
ASTERISK-15365: Asterisk Crashes when a fax comes in over nvfaxdetect
ASTERISK-15366: rtp.c:2482 ast_rtcp_write_sr: rtcp halted Operation not permitted
ASTERISK-15367: [patch] Transferee can hear silence on attended transfer when tranferer hangs up (MOH stops to play)
ASTERISK-15368: Asterisk causes crosstalk between inbound and random channels
ASTERISK-15369: depreciated minmessage still referred to in warning
ASTERISK-15370: gsm to ulaw transcoding sounds terrible
ASTERISK-15371: Segfault while setting up T.38 fax reception
ASTERISK-15372: segmentation fault
ASTERISK-15373: Queue with wrapuptime "call" agent that shouldn't have any call
ASTERISK-15374: No CDR record for non-bridged outgoing calls
ASTERISK-15375: [patch] Optimize queries to cache matches
ASTERISK-15376: [regression] MixMonitor stops recording after transfer using AUDIOHOOK_INHERIT
ASTERISK-15377: When hanging up a channel running chanspy, chanspy does not exit
ASTERISK-15378: Chanspy cannot spy on a non-bridged channel
ASTERISK-15379: [patch] Cannot spy on channel when a local channel is involved
ASTERISK-15380: [patch] Support for SonyEricsson T6x0 and friends is broken
ASTERISK-15381: Verbose for Call Parking is incorrect
ASTERISK-15382: [patch] Bridge application fails when both channels have a similar name
ASTERISK-15383: [patch] sin_family not set to AF_INET when running trunk on Solaris nevada
ASTERISK-15384: Wrong Caller ID on Sip Channel when have a lot of hanging up call
ASTERISK-15385: [regression] chan_local audio crash
ASTERISK-15386: call answer macro not being called in cmd Dial
ASTERISK-15387: [patch] Realtime is broken, blank strings aren't valid any more
ASTERISK-15388: mixmonitor CLI command is broken
ASTERISK-15389: ${BLINDTRANSFER} not set upon transfer
ASTERISK-15390: Updating field in realtime queue table does not take effect
ASTERISK-15391: BlackBerry 7290 will not connect
ASTERISK-15392: memory leak in confic.c
ASTERISK-15393: [patch] call pickup can pickup caller instead of callee
ASTERISK-15394: [regression] Asterisk says "No compatible codecs, not accepting this offer!" on T.38 offer
ASTERISK-15395: Dialout from Meetme conference
ASTERISK-15396: chan_unistim randomly crashes
ASTERISK-15397: Local channel not terminated (regression)
ASTERISK-15398: (local_queue_frame): Error obtaining mutex: Invalid argument (causes crash)
ASTERISK-15399: [patch][regression] Macro executes "h" extension instead of exiting
ASTERISK-15400: Conference Calls
ASTERISK-15401: [patch] TestServer application does not wait long enough after sending its last DTMF digit
ASTERISK-15402: "minute" sound file missing. used by app_queue
ASTERISK-15403: Chan_dahdi calls keep on increase until all the dahdi channel full
ASTERISK-15404: [patch] Huge memory leak
ASTERISK-15405: [patch] Manager interface 'Masquerade' event doesn't include Unique id fields
ASTERISK-15406: [patch] Asterisk crashes in ast_rtcp_write at rtp.c:3536
ASTERISK-15407: [patch] Asterisk produces malformed email files for voicemail
ASTERISK-15408: app_mp3 and chan_local fail
ASTERISK-15409: [patch] Duration and Billsec Decimal Place
ASTERISK-15410: [regression] Voicemail message not recording when voicemail.conf format=wav|gsm|wav49
ASTERISK-15411: Ignores bindaddr on reload
ASTERISK-15412: Error parsing format= parameter in voicemail.conf
ASTERISK-15413: [patch] "config reload" doesn't work correctly
ASTERISK-15414: crash: in "scheduled_destroy" at chan_iax2.c:1511
ASTERISK-15415: Contact header port ignores transport when using externip
ASTERISK-15416: Always get network congestion on second group using .call file
ASTERISK-15417: [patch] [regression] Voicemail information is repeated
ASTERISK-15418: res_fax-1.6.1.5_1.1.6 doesn't trigger T.38 reinvites on fax send (receive works)
ASTERISK-15419: [patch] ParkAndAnnounce() Does Not Seem To Respect Multiple Parking Lots
ASTERISK-15420: ISDN to SIP doesn't generate SIP 180 Ringing with Call Progress ISDN message
ASTERISK-15421: SayUnixTime plays nothing if say.conf mode=new and a format is specified
ASTERISK-15422: [patch] LD (llinker) options not used by main/ and channels/ builds
ASTERISK-15423: app_jack fails to connect to jackd
ASTERISK-15424: ooh323 does not support h245alphanumeric dtmf mode
ASTERISK-15425: [patch] Add Calling and Called Subaddress to CDR record
ASTERISK-15426: app_sms hangs on a call sending an sms
ASTERISK-15427: [patch] astcanary does not exit when asterisk 1.6.2 dies => reopen of #14538
ASTERISK-15428: [regression] I got warnings on remote server when transmitting iax variables
ASTERISK-15429: Asterisk does not send "183 Session Progress" when dialing through a dahdi analog line
ASTERISK-15430: chan_iax2.c dead lock at chan_iax2.c:2563
ASTERISK-15431: [patch] ast_event_cmp always return 1.
ASTERISK-15432: [patch] Deadlock on &(&channels)->lock
ASTERISK-15433: Record() doesn't produce recorded file on hangup with 'x' option
ASTERISK-15434: [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller
ASTERISK-15435: [patch] [OpenSolaris] wav format produces garbage files
ASTERISK-15436: CDR and ForkCDR write wrong data on hangup in AGI execution
ASTERISK-15437: Voicemail subscribtion error
ASTERISK-15438: [patch] option p added to PickupChan to enable pickup a ringing phone by specifying the peer name
ASTERISK-15439: [patch] Group Variables
ASTERISK-15440: Added ability to send DTMF from ExternalIVR
ASTERISK-15441: [patch] Announce to user when they have been muted/unmuted from the AMI
ASTERISK-15442: [patch] Additional information for MeetmeJoin
ASTERISK-15443: [regression] pbx_config does not load when it should
ASTERISK-15444: [patch] Complile issue with h323plus 1.21.0 and pwlib 2.4.5
ASTERISK-15445: asteriskgui not be accessable and softohone
ASTERISK-15446: [patch] overlap dial should terminate when "ast_matchmore_extension" is false
ASTERISK-15447: [patch] ExtensionState should resolve dynamic hints
ASTERISK-15448: [regression] soxmerge arguments broken if Monitoring to absolute path
ASTERISK-15449: RFC2833 DTMF is not passed correctly when going SIP->IAX2->SIP
ASTERISK-15450: Chanspy application does not exit when user hangs up
ASTERISK-15451: No audio is passed from MOH when using originate to a remote peer
ASTERISK-15452: [regression] Originate not launching secondary channel when primary is a Local channel
ASTERISK-15453: [regression] Transfer is broken
ASTERISK-15454: This is a test issue. There's nothing to see here. Please move along.
ASTERISK-15455: This is a 2nd test issue. There's nothing to see here. Please move along.
ASTERISK-15456: [patch] chan_misdn does not set INVALID_EXTEN
ASTERISK-15457: asterisk crashes while fax sending
ASTERISK-15458: macro-hangup executes after 15 minutes and 30 seconds on outbound calls
ASTERISK-15459: MeetMe option 'x' is broken
ASTERISK-15460: [patch] Dial option 'L' does not work correctly when a local channel is involved
ASTERISK-15461: [patch] Update CDR variables that are available, before pbx starts
ASTERISK-15462: Crash In chan_local in local_queue_frame (ast_mutex_trylock)
ASTERISK-15463: [patch] Extend the max number of callgroups/pickupgroups
ASTERISK-15464: [patch] main/feature: additional parking lots not reading needed variables
ASTERISK-15465: [patch] Support for GROUP_MATCH_COUNT regex matching on category
ASTERISK-15466: [patch] Setting "timerb" on chan_sip.conf doesn't work at all, in [general] or peer
ASTERISK-15467: [patch] directmediapermit/directmediadeny support to restrict which peers can do directmedia based on ip address
ASTERISK-15468: [patch] Its not possible to pass more than one agrument in custom features.
ASTERISK-15469: [patch] Session failure with specific SDP-Content (one media specific c= line, no session specific c= line)
ASTERISK-15470: [regression] CDR attended transfer missing
ASTERISK-15471: Asterisk don't update LDAP user's status
ASTERISK-15472: If the UAC execute a SIP registration or deregistration, the LDAP settings don't change.
ASTERISK-15473: [Asterisk 1.6.0-rc 6 update LDAP entries with "null" values]
ASTERISK-15474: Possible problem with IAXVAR
ASTERISK-15475: [patch] Double fields in SQL query
ASTERISK-15476: Segfault under 1.4.23.2
ASTERISK-15477: Transfer hear silence when transfered is busy
ASTERISK-15478: [patch] func_math MATH off by one's
ASTERISK-15479: [patch] [regression] [patch] chan_sip does not check other mailboxes on AST_EVENT_MWI
ASTERISK-15480: [patch] DISA doesn't honor caller ID on the channel
ASTERISK-15481: Getting kernel: asterisk[4278]: segfault at 40 ip 006e6626 sp b70d7e38 error 6 in libc-2.9.so[66d000+16e000]
ASTERISK-15482: RTP Timeout is flawed
ASTERISK-15483: [patch] Random DTMF duplicate emulation on bridged OOH323 channel on outgoing calls
ASTERISK-15484: DTMF not detected at all from Sipgate despite TCPDump showing keypresses
ASTERISK-15485: [patch] Check on ac_cv_pthread_once_needsbraces fails
ASTERISK-15486: ms sql connections problems
ASTERISK-15487: Asterisk 1.6.0-rc 6 update LDAP entries with "null" values
ASTERISK-15488: callee chanel overwrites the caller cdr
ASTERISK-15489: Asterisk manage forked calls as reinvite
ASTERISK-15490: [patch] TLS socket file descriptor fails to open (with no error message in log)
ASTERISK-15491: [patch] channels stuck in ringing state forever
ASTERISK-15492: "sip show peer/user <tab>" doesn't complete correctly
ASTERISK-15493: 1.6.2.1: sip BYE issued 90 seconds into call
ASTERISK-15494: [patch] deadlock in app_queue with use_weight during reload
ASTERISK-15495: [patch] segfault on chanspy due to race in main/channel.c
ASTERISK-15496: [patch] res_phoneprov.so causes Asterisk to crash on ${MAC}-phone.cfg file
ASTERISK-15497: [regression] .call file not connecting to context: when channel: answers
ASTERISK-15498: Hundreds (thousands?) of WARNING messages when data sent via res_phoneprov
ASTERISK-15499: [patch] warning about "Invalid peer port configuration" for realtime
ASTERISK-15500: [patch] AEL2 parser messages missing a final \n
ASTERISK-15501: [patch] Configured CFLAGS/LDFLAGS are used by main, but not by modules built out of tree
ASTERISK-15502: [patch] app_dial does not respect GOSUB_RESULT
ASTERISK-15503: [patch] app_dial gosub does not pass back GOSUB_RETVAL
ASTERISK-15504: [patch] Console documentation not loaded from XML
ASTERISK-15505: [patch] Crash in res_agi when trying to send application usage
ASTERISK-15506: chan_mobile crashes asterisk segmentatio fault on end of outgoing call
ASTERISK-15507: [patch] new feature T.38 switch on/off from dialplan
ASTERISK-15508: [patch] Missing fallback to audio fax feature when T.38 re-INVITE failed for 1.4
ASTERISK-15509: [patch] There is an Active call, even though device is Unregistered from asterisk!
ASTERISK-15510: [regression] Voicemail admin only records .wav and .gsm, not .WAV greetings/unavailable/temporary messages
ASTERISK-15511: [patch] New AgentTransfer manager event with extended transfer information
ASTERISK-15512: [patch] Solaris sed fails on generating ael_lex.c
ASTERISK-15513: IAX always attempts authentication against first (alphabetically) user
ASTERISK-15514: gotoiftime does not work as expected for date range
ASTERISK-15515: Segmentation Fault
ASTERISK-15516: [regression] Attended transfer broken in 1.6.1.13
ASTERISK-15517: T.38 fix missing
ASTERISK-15518: Lags when using imap
ASTERISK-15519: [patch] [regression] 1.6.2.7 hangs during initial module load on Darwin
ASTERISK-15520: [patch] [regression] T.38 negotiation Broken
ASTERISK-15521: SIP URIs are not always parsed correctly
ASTERISK-15522: [patch] Caller name lost during call redirect
ASTERISK-15523: [patch] Introduce function for parsing ABNF name-andor-addr = name-addr / addr-spec
ASTERISK-15524: [patch] Added a config parameter to report span and/or channels alarms using AMI
ASTERISK-15525: [patch] [regression] DTMF Relaying appers broken
ASTERISK-15526: Billing Differences between Carrier and Asterisk
ASTERISK-15527: [patch] CHANNEL function cannot set OSP token for outbound IAX calls.
ASTERISK-15528: [patch] Build issues with FreeBSD 6/8
ASTERISK-15529: [patch] PRI locks randomly, hangup cause 102, "recovery on timer expiry".
ASTERISK-15530: 'interval' option doesn't work
ASTERISK-15531: Orinate calls using AMI on version 1.4.29 broken.
ASTERISK-15532: Unable to link Voicemail to voicemail accounts created using MySQL
ASTERISK-15533: realtime oracle engine
ASTERISK-15534: Every time I type "odbc show" it crashes in the next few seconds
ASTERISK-15535: after a few minutes it takes down the server
ASTERISK-15536: [patch] app_queue: Give members a penalty time for not answering
ASTERISK-15537: type=user and type=friend are no longer the same for chan_sip
ASTERISK-15538: coredump on T.38 Session with 1.6.2.1
ASTERISK-15539: [patch] Add support for configurable peer username in digest authentication
ASTERISK-15540: I can't store CDRs in mysql DB
ASTERISK-15541: Timezone (tz) parameter won't apply for users.conf
ASTERISK-15542: [regression] Local channels broken for Originate and .callfiles : Call failed to go through, reason (3) Remote end Ringing
ASTERISK-15543: configure --with-netsnmp fails if no openssl-devel
ASTERISK-15544: [patch] core dump when user parkandannouce
ASTERISK-15545: [patch] Add AMI support for device states
ASTERISK-15546: Setvar DEVICE_STATE manager action occasionally ignored.
ASTERISK-15547: Core Dump on Exit - v1.6.2/FreeBSD
ASTERISK-15548: [patch] realtime oracle engine
ASTERISK-15549: Callcentric Dropping Registration in Version 1.6.2 [see issue 0012312 ]
ASTERISK-15550: large memory leak - ast_rtp_destroy not being call in all circumstances/other associated memory not being freed from chan_sip
ASTERISK-15551: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.2.37.tar.gz does not contain issue# 0015765
ASTERISK-15552: [patch] New mute feature for MixMonitor
ASTERISK-15553: Application Directory does not respect the [vm-context] context and selects users from the default context
ASTERISK-15554: [patch] Missing description of the PARKINGLOT variable in XML documentation
ASTERISK-15555: [patch] 'moh reload' doesn't reload moh directory content
ASTERISK-15556: BASE64_DECODE broken again
ASTERISK-15557: Caller ID info is destroyed after FXS channel is ringed
ASTERISK-15558: Inbound calls are dropped after 15 mins and several Status 400 and 422 messages in SIP trunk against Huawei SoftX3000
ASTERISK-15559: [patch] DSP progress detection unable to detect SIT
ASTERISK-15560: Audio loop reports T38 switchover but t38state != T38_STATE_NEGOTIATED
ASTERISK-15561: [patch] Initial pause not implemented, but documented as available.
ASTERISK-15562: IAX2 Reject Not Shown in Debug
ASTERISK-15563: [patch] Clean transmit_* for start/stop media transmission
ASTERISK-15564: [patch] Multiple segfaults in leave_voicemail at app_voicemail.c:4451 Asterisk 1.4.29
ASTERISK-15565: E1 PRI channel 'glare', where asterisk hangups up the inbound call from network.
ASTERISK-15566: app_fax doesn't receive fax with T.38
ASTERISK-15567: [patch] Parking a call, then retrieving it with ParkedCall() kills the ability to transfer the retrieved call.
ASTERISK-15568: [patch] zero/empty argument to gosub yields callers $ARG1
ASTERISK-15569: Attended transfers get incorrect voicemail.
ASTERISK-15570: [patch] Adding manager event JabberStatus
ASTERISK-15571: Sip Channels Colapse
ASTERISK-15572: [patch] SIP call documentation - feel free to edit
ASTERISK-15573: [patch] T.38 negotiation fails with Patton SN2400
ASTERISK-15574: [patch] Deadlock between handle_request_do and do_devstate_changes
ASTERISK-15575: SIP Message parameters and URI parameters not parsed correctly
ASTERISK-15576: [patch] Send manager event on AMI command Bridge
ASTERISK-15577: [patch] Unrecognized prilocaldialplan NPI modifier
ASTERISK-15578: T.38 with devices behind NAT does not work
ASTERISK-15579: [patch] After AMI Bridge action the callerid's on the phones are not updated.
ASTERISK-15580: Background application does not return until file is finished being played (Asterisk 1.6.0.10)
ASTERISK-15581: [patch] [regression] 1.6.1.13 and 1.6.1.14 UDP ports not freed
ASTERISK-15582: Queue with autofill=no and strategy=ringall sometimes rings non-oldest caller through to agents
ASTERISK-15583: app_queue does not change member state
ASTERISK-15584: [patch] Calendar SIGSEGV with iCal
ASTERISK-15585: Calls retrieved from parking lot not recorded
ASTERISK-15586: [patch] Pause After call
ASTERISK-15587: make install fails when GNU install missing
ASTERISK-15588: [patch] res_pktccops.so doesn't export a symbol, chan_mgcp will not load or will malfunction depending on gcc version
ASTERISK-15589: [patch] 99.9 cpu when asterisk started with init.d script
ASTERISK-15590: Park ed call slot annoucement is not heared
ASTERISK-15591: [patch] Remove coloring escape sequences from log files.
ASTERISK-15592: Asterisk drop calls sending a CSeq: 103 BYE
ASTERISK-15593: [regression] Realtime agents Device state always Not in use
ASTERISK-15594: [patch] Overlap receiving timeout, plus dialplan latency, causes network to retry SETUP
ASTERISK-15595: [patch] 606 Not Acceptable is also a valid response to reject a T.38 re-INVITE
ASTERISK-15596: T.38 session fails with '488 Not acceptable here' if within 5 seconds there is no "SIP 100 Trying" REINVITE reply from remote
ASTERISK-15597: 1.4 does not send any SIP messages after the "100 Trying" to the T.38 INVITE requesting side
ASTERISK-15598: Crash when destroying sip channels
ASTERISK-15599: Originating a call from within the dialplan using Originate() does not result in a CDR
ASTERISK-15600: [patch] Update DUNDi to XML docs
ASTERISK-15601: astobj2.c:279
ASTERISK-15602: problems with high call frequency and more than 300 calls at the same time
ASTERISK-15603: [patch] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
ASTERISK-15604: No Branch 1.6.1 Patch for Issue #0016766
ASTERISK-15605: [patch] ParkAndAnnounce Core dumps asterisk
ASTERISK-15606: [patch] segfault in pri_schedule_del at prisched.c:124
ASTERISK-15607: [patch] make sounds doesn't download but make install does
ASTERISK-15608: Create example documentation for usage of FILTER() to help secure dialplans
ASTERISK-15609: [patch] Design functionality test for dialplan pattern matching
ASTERISK-15610: Dialplan language does not deal with & character safely
ASTERISK-15611: Heavy locking in manager.c results in eventual crash and loss of CLI commands and intense CPU load
ASTERISK-15612: dfgfdg
ASTERISK-15613: Failure of canary
ASTERISK-15614: [patch] chan_sip does not decrease module refcount on deferred BYE
ASTERISK-15615: [patch] Attended transfer broken in 1.6.2.2
ASTERISK-15616: [patch] Reload command does not update the SLA configuration properly
ASTERISK-15617: [patch] Manager event AgentComplete should alway be sent
ASTERISK-15618: Parameter m in Dial command
ASTERISK-15619: SDP in a session refresh re-INVITE does not contain T.38 when it should
ASTERISK-15620: IAX2 queue member Unknow instead of Not in Use
ASTERISK-15621: (Version 1.4.29) Queue with autofill=no and strategy=ringall sometimes rings non-oldest caller through to agents
ASTERISK-15622: [patch] 603 Declined when call torn down
ASTERISK-15623: Discrepancy among core-sounds-en.txt and sounds in 1.4.17 set
ASTERISK-15624: [patch] Endianess problems in skinny messages
ASTERISK-15625: Asterisk does not honor the bindport.
ASTERISK-15626: [patch] DEBUG_THREADS - Compile errors on OS/X Snow Leopard
ASTERISK-15627: Audio pauses at the same time as rtcp report is handled
ASTERISK-15628: [patch] Build fails on smsq.c with syntax error
ASTERISK-15629: Random crashes
ASTERISK-15630: [patch] [regression] autofill=no always IGNORED.
ASTERISK-15631: Asterisk 1.6.2.3 RC2 CRASHING RANDOMLY
ASTERISK-15632: seg fault in _ast_calloc at utils.h:462
ASTERISK-15633: [patch] core show sysinfo shows invalid (negative value) for ram on systems whith a lot of ram
ASTERISK-15634: freeplay and opsound tarballs are identical
ASTERISK-15635: ASterisk 1.6.0.13
ASTERISK-15636: [patch] Deadlock in chan_local when obtaining locks on local_pvt->lock
ASTERISK-15637: Codec translation path builder does not produce expected results with 16khz and 32khz audio
ASTERISK-15638: [patch] [regression] system() dialplan function fails (-1 returned) when argument is single quoted
ASTERISK-15639: [patch] Fix portability bit fields and make "mfcr2_immediate_accept" work again
ASTERISK-15640: Incorrect linker flags used on OpenSolaris
ASTERISK-15641: Unable to create channel for non registered SIP devices
ASTERISK-15642: [patch] Deadlock between dahdi_exception and dahdi_indicate
ASTERISK-15643: [patch] Voicemail attachments are sent even with attach=no
ASTERISK-15644: Should there be transcoding after attended transfer?
ASTERISK-15645: CRASH ON 1.6.2.3 RC2
ASTERISK-15646: [patch] Some issues with New SDP handling code and T.38
ASTERISK-15647: [patch] Memory leak in realtime meetme
ASTERISK-15648: [regression] Crash in app_voicemail.c in function inprocess_cmp_fn becuase j->context is NULL
ASTERISK-15649: Template examples in documentation imply well defined overriding semantics, but this is not true
ASTERISK-15650: [regression] Blind transfers initiated from calling party aren't disconncted
ASTERISK-15651: Incorrect checking of Refer-To and Referred-By SIP headers
ASTERISK-15652: [patch] asterisk command history loads as unusable garbage
ASTERISK-15653: Music on hold broken in 1.6.2.2
ASTERISK-15654: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
ASTERISK-15655: [patch] [regression] One-legged Transfer (INVITE / Replaces) not working anymore
ASTERISK-15656: [patch] PickupChan is not working
ASTERISK-15657: [patch] Voicemail minsecs is not able to be overridden per-mailbox
ASTERISK-15658: [regression] No Ringing / No Playback when call is getting forwared by a phone
ASTERISK-15659: [patch] Set conterence options on pin-less RealTime conferences
ASTERISK-15660: Phone generated dial tone not switched off during call
ASTERISK-15661: One way audio after placing call on hold and resuming
ASTERISK-15662: [patch] Make verb and noun declination available in dial plan
ASTERISK-15663: [regression] AGI function GET DATA does not play audio file
ASTERISK-15664: codec_dahdi.c gives error message on startup after fresh install
ASTERISK-15665: Calltoken's and IAX2 realtime configuration
ASTERISK-15666: [patch] Cleanup transmit_* functions
ASTERISK-15667: [patch][regression] Cannot specify http:// URL in music-on-hold class directory statement
ASTERISK-15668: setting pollmailboxes=yes will crash asterisk on startup
ASTERISK-15669: ael context extend does not look at macros
ASTERISK-15670: [patch] Cleanup transmit_displaymessage
ASTERISK-15671: [patch] Enable opening an audio stream post-startup
ASTERISK-15672: [patch] unable to exit echo application with '#'
ASTERISK-15673: Asterisk sends voicemails to a second totally wrong receiver
ASTERISK-15674: Avaya IP Office 5.0 probably crashes Asterisk
ASTERISK-15675: Asterisk 1.6.1.16 crashes randomly
ASTERISK-15676: [patch] CLI command 'core show codecs' does not display slin16 codec 0x8000
ASTERISK-15677: Attended transfer is broken on 1.6.2.4
ASTERISK-15678: /etc/init.d/asterisk.debian consumes 100% CPU
ASTERISK-15679: Sip Channels Colapse - Too much sip channels
ASTERISK-15680: Asterisk 1.2.39 core dump
ASTERISK-15681: [regression] Context option in queue.conf no longer works
ASTERISK-15682: IAX2 crash's randomly
ASTERISK-15683: queuelog does not show correct extension on transfers using non Local/ members
ASTERISK-15684: [patch] Realtime queue does not re-read announce variable from mysql after first use
ASTERISK-15685: Repark a call on Parkinglot
ASTERISK-15686: Asterisk Crashes after it thinks it gets corrupt SIP message
ASTERISK-15687: [patch] From-header parsed twice for each invite or subscription request
ASTERISK-15688: dnsmgr failes to match peer when SIP srvlookup is on
ASTERISK-15689: srvlookup don't work with register
ASTERISK-15690: [patch] patch for LISTFILTER to properly handle delimiter
ASTERISK-15691: Asterisk 1.4.29 musiconhold of remote party doesnt work
ASTERISK-15692: Compile fails in dahdi_show_status with kernel 2.6.33
ASTERISK-15693: [patch] Incorrect pattern specificity in new dial pattern functions
ASTERISK-15694: [patch] [regression] Duplicate TXREQ packets will cause chan_iax2 to reject an unrelated call in the future
ASTERISK-15695: [patch] system() dialplan function does not work
ASTERISK-15696: [patch] moh files install under datadir, at runtime: under varlibdir
ASTERISK-15697: jerky audio after a while
ASTERISK-15698: [patch] SIP autocreate peers registered when request to unregister
ASTERISK-15699: [patch] Useful new wildcards to ease secure dialplans
ASTERISK-15700: [patch] Alignment trap on ARM processor on calculating cost of codec
ASTERISK-15701: crash in musiconhold
ASTERISK-15702: Adaptive jitterbuffer causes 30 seconds of no audio.
ASTERISK-15703: jblog=yes does not create jblog like expected
ASTERISK-15704: regcontext not handled in a fashion similar to chan_sip
ASTERISK-15705: Allow regcontext per peer
ASTERISK-15706: [patch] VoiceMail(vmbox@context,s) -> Regularly segfaults asterisk
ASTERISK-15707: Duplicates of uniqueID
ASTERISK-15708: RTP traffic only seen in one direction when using FollowMe()
ASTERISK-15709: t38pt_usertpsource=yes seems to work incorrectly with ReceiveFax
ASTERISK-15710: [patch] app_queue: Log failed attempts to call members
ASTERISK-15711: Add support for reporting/passing all CallerID data to app_queue - specifically RDNIS,DNID
ASTERISK-15712: [patch] Segfault branches, and trunk, when DAHDI FXS port goes off hook
ASTERISK-15713: [patch] The Dial c option returns answered elsewhere if the dial timeout occurs (only tested using SIP)
ASTERISK-15714: [patch] Patch to fix 15609 broke followme
ASTERISK-15715: [patch] [regression] app_followme playing wrong sound files
ASTERISK-15716: mutual exclusion with optiom m and L in dial
ASTERISK-15717: unexpected end of a call for 20-25 seconds
ASTERISK-15718: [patch] [sounds] Two additional sound prompts for use with ConfBridge
ASTERISK-15719: glibc 2.11.1 causes asterisk to not start due to return code from dlclose
ASTERISK-15720: Buggy parse of request-line in function check_user_full()
ASTERISK-15721: [patch] Qualify frequency has big pauses. Asterisk stops sending SIP OPTIONS to keep NAT alive
ASTERISK-15722: For loop never exists when calling an extension that exists with BUSY, failure to leave 'for' loop in 'scan_thread'
ASTERISK-15723: [patch] Cleanup transmit_callstate handling
ASTERISK-15724: [patch] Problem inserting CDR records when certain characters are used
ASTERISK-15725: SIP RTP audio delay
ASTERISK-15726: [patch] chan_mgcp crash Adtran 624 asterisk 1.8.0 Beta 2
ASTERISK-15727: 100% CPU load at feature startup
ASTERISK-15728: Need to port a feature from Trunk to 1.6.X
ASTERISK-15729: Monitor() does not handle filenames with path correct
ASTERISK-15730: [patch] fix getting callerid name in imap_retrieve_file() (broken callerid number announcement/reply/...)
ASTERISK-15731: Call that clears in same app_dial poll as answer is reported as NOANSWER but NORMAL_CLEARING
ASTERISK-15732: app_fax exits with returncode -1, transmission error, although fax is sent correctly, AGI aborts execution
ASTERISK-15733: Callerid Channel dahdi port FXS hook up
ASTERISK-15734: sip reinvite broken
ASTERISK-15735: All extensions are patterns (despite it does not begin with _ (underscore) if extenpatternmatchnew=yes
ASTERISK-15736: It crashes on func_odbc
ASTERISK-15737: [patch] [regression] SQL Syntax Error - Missing Single Quote
ASTERISK-15738: [patch] [regression] local channel with '/bn' modifier does not optimize itself on built-in transfer
ASTERISK-15739: In dialplan (extensions.conf), subdirectories are not respected
ASTERISK-15740: [patch] Rogue Newchannel events for failed Originate calls
ASTERISK-15741: crash in main/cli.c: find_cli
ASTERISK-15742: SIP-Provider without "SIP/2.0 180 Ringing" makes trouble with call file
ASTERISK-15743: Read factory 0xb6d0acb8 was pretty quick last time, waiting for them
ASTERISK-15744: UserEvent Documentation is incorrect
ASTERISK-15745: [patch] Automatic add UniqueID to user event
ASTERISK-15746: [patch] Update to new local channel documentation
ASTERISK-15747: When transfering call, sound disappear.
ASTERISK-15748: [patch] DBGet response does not end with a 'Complete' event
ASTERISK-15749: SpeechBackground with multiple files does not interrupt speech when DTMF is received
ASTERISK-15750: Nested Dial()s that use U() or M() results in: '&(audiohook)->lock' freed more times than we've locked!
ASTERISK-15751: [patch] Callerid Channel dahdi port FXS are empty after the first hangup.
ASTERISK-15752: Catch privateNumberDigits from h323 pdu
ASTERISK-15753: Using func_odbc.conf
ASTERISK-15754: When playing from ExternalIVR, the playback is very fast (about double the speed of standard Playback)
ASTERISK-15755: [patch][feature] Add device state capabilities to ConfBridge (similar to MeetMe)
ASTERISK-15756: multiple values for "dtmfmode=" per trunk
ASTERISK-15757: Deadlocks with ~2k MGCP users
ASTERISK-15758: [patch] OOH323 connection to Avaya IPOffice 403 drops i n 394 seconds
ASTERISK-15759: After upgrade from 1.4.21 to 1.4.29 on internal SIP calls don't hear the ringback tone
ASTERISK-15760: cli help for alias doesnt do anything sensible
ASTERISK-15761: contributed init.d file has wrong pid location
ASTERISK-15762: [patch] Cleanup transmit_ for handle_register and keepalives
ASTERISK-15763: [patch] MusicOnHold ignores realtime when no musiconhold.conf classes configured.
ASTERISK-15764: alarm state not properly maintained on analog channels
ASTERISK-15765: impossible to pass more than one argument to a custom applicationmap in features.conf
ASTERISK-15766: Asterisk segmentation fault
ASTERISK-15767: ${VOICEMAIL_EXTEN} doesn't get set at asterisk startup
ASTERISK-15768: [patch] small error in T.140 RTP port verbose
ASTERISK-15769: When attempting to dial specific number pattersn a CHANUNAVAIL message is received
ASTERISK-15770: When Answer is used for chan_local in Originate, the originate go crazy
ASTERISK-15771: asterisk 1.6.2.6-rc2 has core show locks where DEBUG_CHANNEL_LOCKS is not set
ASTERISK-15772: [patch] Can't build asterisk using just --with-netsnmp... seems to always want to use CONFIG_NETSNMP
ASTERISK-15773: incoming INVITE received no progress, just 200 OK, causing Sipra pstn to go off-hook
ASTERISK-15774: [patch] MusicOnHold produces a crash
ASTERISK-15775: [patch] Cleanup transmit_* functions
ASTERISK-15776: [patch] Crash in app_voicemail.c in function retrieve_file (Read out in small chunks)
ASTERISK-15777: Audio problems in Meetme when DAHDI channel is Monitored
ASTERISK-15778: [patch] Chan_sip compile never completes on Mac OSX 10.6.2
ASTERISK-15779: Transfer fails
ASTERISK-15780: [patch] t38pt_udptl=no on a peer isn't respected (can't disable t38 negotiation on a per-peer basis)
ASTERISK-15781: Bug in Calculating T.38 far max IFP?
ASTERISK-15782: [patch] [regression] Segfault when hanging up phone after launching app_confbridge on Solaris 10 x86
ASTERISK-15783: [patch] sqlite module does bad parsing of values in config file
ASTERISK-15784: Crash when using an SQL Server ODBC connection (FreeTDS)
ASTERISK-15785: [regression] Set(CDR(mycol)=xxx) does not get populated in the backend when the CDR is posted
ASTERISK-15786: Data Buffer Size Exceeded!
ASTERISK-15787: [patch] Asterisk sends session-timer with "require" after 15 minutes
ASTERISK-15788: [patch] func_odbc query is limited to 15 characters
ASTERISK-15789: [patch] trivial patch to notifiy CLI when module loaded / unloaded.
ASTERISK-15790: [patch] RTP Timestamp changes after transfer, but SSRC not and the markerbit ist not set.
ASTERISK-15791: [patch] Autopause only pauses member on queue where timeout took place.
ASTERISK-15792: Dialplan continues execution after transfer
ASTERISK-15793: When a context is defined in [general] section of sip.conf, other contexts are ignored
ASTERISK-15794: Context statement removed from [general] section of sip.conf, but remained active even after a reload
ASTERISK-15795: [patch] endless wait for RTP in certain scenarios
ASTERISK-15796: ./configure -help describes wrong default behavior
ASTERISK-15797: Can't receive Fax with Patton 4554 ISDN Gateway and SPA 2102 VoIP Adapter
ASTERISK-15798: Saturated Handles, Socket Error
ASTERISK-15799: [patch] [regression] videosupport=always acts like videosupport=no
ASTERISK-15800: SQLite3-3.6.23 incompatibility in asterisk-1.6.2.6
ASTERISK-15801: [patch] Background application not use 'context' parameter
ASTERISK-15802: SIP response 415 "Unsupported Media Type" when using G729
ASTERISK-15803: On omitting the T flag from Dial() the caller can still make a blind transfer
ASTERISK-15804: [patch] Exchange Web Service calendaring support
ASTERISK-15805: [regression] Manager Events Exit Early
ASTERISK-15806: Park is passed incorrect parameters if a Dial application is executed in a subroutine anywhere on the system.
ASTERISK-15807: [patch] This is C. Indent levels do not matter in C.
ASTERISK-15808: Asterisk crashes with app_fax / spandsp when compiled in 64 bit
ASTERISK-15809: [patch] Makefile update to only build asterisk.conf
ASTERISK-15810: [patch] make clean: /bin/sh: /usr/bin/sw_vers: not found
ASTERISK-15811: 181 forwarded messages incorrectly interpreted
ASTERISK-15812: Crash with 'sip reload' when system has some load
ASTERISK-15813: [patch] Makefile: remove ASTBINDIR variable
ASTERISK-15814: When using originate Local/.../n, dest extension does not run on Local channel pickup
ASTERISK-15815: Jabber Module, crash and/or no authentication
ASTERISK-15816: [patch] Problems with MeetMe and RT schedule dates
ASTERISK-15817: Conflict in sample configurations with TRUNK, etc.
ASTERISK-15818: [patch] Perl script to import CDR text file to ODBC database table
ASTERISK-15819: [patch] internal_ao2_ref fails to check if null returned from INTERNAL_OBJ
ASTERISK-15820: [patch] Insert fails when database initialized during connection outage to postgres server
ASTERISK-15821: [patch] allow using system copy of libedit
ASTERISK-15822: [patch] Explicit context set in SIP peer overridden by default domain context
ASTERISK-15823: [patch] Documentation for configuring asterisk faxdetect for fax to email with spandsp
ASTERISK-15824: Create CDR queue so records are not lost when connectivity is lost
ASTERISK-15825: dtmf logging to console not working
ASTERISK-15826: [regression][patch] SDP c and o lines contain the wrong IP address when using an externally mapped IP(extern{ip,host})
ASTERISK-15827: Configure script inconsistent in using CFLAGS when detecting header files
ASTERISK-15828: [patch] Add new AGI command: PARK
ASTERISK-15829: [regression] Some AMI Originate Calls on chan_local fail/timeout
ASTERISK-15830: [regression] Incoming DAHDI to DAHDI bridged channels staying with ringing status
ASTERISK-15831: [patch] CLI commands via asterisk -rx may not return all output
ASTERISK-15832: channel.c:2753 __ast_read: Exception flag set on 'SIP/XXXXXXXX-000000e3', but no exception handler
ASTERISK-15833: main/test.c reports erroneous cli message
ASTERISK-15834: chanspy on channel in MeetMe conference results in a crash
ASTERISK-15835: Asterisk crash - core dump at manager.c:2637
ASTERISK-15836: strange documentation of tlsbindaddr in sip.conf
ASTERISK-15837: timeout problem in queues
ASTERISK-15838: [patch] Clear received caller ID number and name on DADHI hangup
ASTERISK-15839: [patch] peer section does not allow to configure a port
ASTERISK-15840: [patch] Freenum-in-a-can configuration for configs/extensions.conf.sample
ASTERISK-15841: What causes 'Bad Magic Number'? As immediately after segfault.
ASTERISK-15842: problem with announce frequence in queues
ASTERISK-15843: Queue hangup if periodic announcement file is missing
ASTERISK-15844: [patch] Meaningless extension warnings logging
ASTERISK-15845: CPU usage increases if WaitEvent not called
ASTERISK-15846: redirect to fax extension only after ring
ASTERISK-15847: [patch] Real-time Priory
ASTERISK-15848: Adaptive Jitter Buffer issue
ASTERISK-15849: [patch] chan_dahdi/fxs really needringing
ASTERISK-15850: AGI->wait_for_digit or AGI->exec('Read' do not report digits back on an outgoing call
ASTERISK-15851: asterisk is not closing unused RTP ports
ASTERISK-15852: When using another SIP Trunk, Asterisk generates the initial ring RING as a response to "SIP SESSION PROGRESS"
ASTERISK-15853: Asterisk 1.6.1.18 and 1.6.2.6 RTP BUG
ASTERISK-15854: [patch] Added response timeout option to SendFax() appliation
ASTERISK-15855: No ENTERQUEUE event in queue_log if leavewhenempty=yes in queues, therefore no ACD report is available to track overflow calls.
ASTERISK-15856: Unable to Install Asterisk in asterisk-1.6.0.21
ASTERISK-15857: [patch] [regression] fix for #16802 forces change of astrundir ownership, breaking socket perms
ASTERISK-15858: [patch] Fix query with double backslash in string literals and stop log warnings
ASTERISK-15859: [patch] MixMonitor records shorter files than the call duration.
ASTERISK-15860: qsigchannelmapping parameter
ASTERISK-15861: [patch] Asterisk crashes while core restart (#0 0x000000000050683c in term_beep (el=0x16cdd9b0) at term.c:865)
ASTERISK-15862: [patch] Memory Leak in app_queue
ASTERISK-15863: [patch] Improve realtime queue logging
ASTERISK-15864: [patch] Asterisk crash in func_odbc
ASTERISK-15865: [patch] [regression] Overlap dialing to PSTN failing after #16789
ASTERISK-15866: [patch] 'core show settings' should show all settable directories
ASTERISK-15867: [patch] Segfault in manager event after fax receipt
ASTERISK-15868: app_followme + cdr_adaptive_odbc crashes when followme progresses from number set to number set
ASTERISK-15869: Background behaves strangely[t when priority 1 is not available in current extension.
ASTERISK-15870: MWI SIP NOTIFY may contain wrong "Via: ..." header, making the phone discard the whole message
ASTERISK-15871: T.38 faxmaxdatagram overflows with UDPFEC, works with "t38pt_udptl=yes,redundancy" (udptl.c)
ASTERISK-15872: [patch] Segmentation fault when using two codec modules that register the same src and dst format
ASTERISK-15873: [patch] AGI SPEECH SET bugs
ASTERISK-15874: unportable shell in safe_asterisk
ASTERISK-15875: On receiving Fax with ReceiveFax Asterisk core dumps
ASTERISK-15876: C keeps ringing when hanging A and B after blind transfer using atxfer
ASTERISK-15877: [patch] Pickup with Aastra phones: Unable to find subscription
ASTERISK-15878: Crash just after call is answered
ASTERISK-15879: [patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
ASTERISK-15880: AMI IAXpeers not "complete," no actionID
ASTERISK-15881: [patch] Asterisk Crashes With Core Dump When FUNC_ODBC Processes SQL Statements With Errors
ASTERISK-15882: [patch] [regression] No message is actually prepended when sending a voicemail message to another user
ASTERISK-15883: 1.6.1.18 -> 1.6.2.6 T38 Fax: call drops
ASTERISK-15884: No progress playing a sound after hangup
ASTERISK-15885: Add a "pickupmacro=" setting in features.conf for execution on undirected pickup (typically *8)
ASTERISK-15886: [patch] [regression] openh323 log is not printed on asterisk console
ASTERISK-15887: [regression] realtime queues: persistent members not loaded from astdb on restart
ASTERISK-15888: MeetMe 'useropts' and 'adminopts' not taken into account when RealTime is used
ASTERISK-15889: Far Max IFP/Local max datagram Calculation and the reality
ASTERISK-15890: ChannelRedirect() fails to properly redirect some of the time
ASTERISK-15891: Segfault error
ASTERISK-15892: When providing a delemeter character, placing it betwenn apostrophy signs does not work. CUT retruns the shole string in this ca
ASTERISK-15893: Socket leaks when SIP call is rejected
ASTERISK-15894: ChannelRedirect() fails to redirect
ASTERISK-15895: Random crashes
ASTERISK-15896: (Regression) Pickup from Grandstream BLF button ignores the context specified in Pickup command
ASTERISK-15897: Music On Hold Not reloading the files using moh reload
ASTERISK-15898: Using pattern match in a hint causes deadlock under described conditions
ASTERISK-15899: [patch] Missing file queue-minute causes hangup of queue calls with wait time announcement = 1 minute
ASTERISK-15900: [patch] Potential malfunction due to unitialized local variable
ASTERISK-15901: [patch] minmemfree does not work
ASTERISK-15902: [patch] Make transfer calls more pattern friendly
ASTERISK-15903: [patch] Softkey redial with no previous number segfaults
ASTERISK-15904: [patch] Wrong encoding of SIP URI
ASTERISK-15905: [patch] endless cycle in ast_waitfor_nandfds() for big timeouts
ASTERISK-15906: Asterisk crashes after "Stopped music on hold on DAHDI" (or tranfer to SIP channel may be)
ASTERISK-15907: voicemail blasting to users with voicemail as an email does not seem to be working
ASTERISK-15908: SENDFAX is not working for me
ASTERISK-15909: Dial 'm' option produces a lot of warnings on DAHDI channel
ASTERISK-15910: When single user in Meetme application there are scrolling errors from ast_dsp_silence
ASTERISK-15911: Asterisk 1.4.30 crashes on transers with the patch around CONNECTEDLINE (https://issues.asterisk.org/view.php?id=8824#118065)
ASTERISK-15912: [patch] Message count incorrect
ASTERISK-15913: [patch] SegFault when connecting incoming call to parked call.
ASTERISK-15914: Conference sound files re-recorded
ASTERISK-15915: [patch] CallerID not properly set when using Originate and AGI
ASTERISK-15916: app_voicemail crashes intermittently when voicemail box is over maxmsg
ASTERISK-15917: Request to change Name/username field truncation from 11 to 12 chars
ASTERISK-15918: [patch] Enable PRI SERVICE message support in chan_dahdi
ASTERISK-15919: hidecalleridname parameter in chan_dahdi.conf
ASTERISK-15920: [patch] Endpoints are not loaded when using Realtime
ASTERISK-15921: AEL warning when using Return() application is misleading
ASTERISK-15922: Extra lines in msg<number>.txt is added when forwarding a prepended mail to another mailbox
ASTERISK-15923: Segfault with too many IMAP voicemails
ASTERISK-15924: Everyone is busy/congested at this time
ASTERISK-15925: Busy(xx) exits immediately on IAX channel
ASTERISK-15926: ENUMQUERY fails for seemingly valid e164.org record
ASTERISK-15927: Playback deadlock?
ASTERISK-15928: [patch] CLI command logger set level auto complete
ASTERISK-15929: chan_sip sends to peer mwi notify for wrong mailbox
ASTERISK-15930: [patch] Add ${TOTALCALLS} dialplan variable
ASTERISK-15931: SIP DTMF problem using RFC2833 between 1.2 <-> 1.4 <-> Unknown-brand/model SBC
ASTERISK-15932: [patch] Command/Response queue stuck
ASTERISK-15933: proxy_allocate() fails to init proxy->ip.sin_family
ASTERISK-15934: MeetMe Options S(10)L(10000)
ASTERISK-15935: [patch] HowTo: Collecting Debug Information
ASTERISK-15936: [patch] CLI prompt interfers with CLI output
ASTERISK-15937: [patch] OSARCH in GNU/kFreeBSD
ASTERISK-15938: missing libs in link command of chan_h323.so: module fails to load
ASTERISK-15939: [patch] Bluetooth linkkey is never stored and as a result phone PIN is requested every time a new connection is started
ASTERISK-15940: [patch] ALL VERSIONS ! between two asterisks and peers authenticate as coincidental name invite
ASTERISK-15941: [branch] update live_ast for testsuite
ASTERISK-15942: Caller ID from internal DAHDI extensions not detected
ASTERISK-15943: RECONNECT fails to work for Conference Call
ASTERISK-15944: [patch] downgrade ast_debug from 1000 to 10.
ASTERISK-15945: [patch] no sound on Playback(<file>,noanswer)
ASTERISK-15946: Specifying return_context for ParkAndAnnounce affects 'h' exten behavior
ASTERISK-15947: [patch] cdr_mongodb
ASTERISK-15948: [patch] sip-friends.sql Missing 'useragent'
ASTERISK-15949: [branch] Appdoc for manager events
ASTERISK-15950: Connection Problem with UnixODBC 2.1.14
ASTERISK-15951: [patch] Updated Mantis Work Flow Documentation
ASTERISK-15952: sip show channelstats isses.
ASTERISK-15953: asterisk 1.4.23 regularly core dumps
ASTERISK-15954: SDP does not get parsed when in SIP multipart body below line 64
ASTERISK-15955: [patch] SetCallerpres not honored on SIP Redirect
ASTERISK-15956: [patch] Add a parameter to SendDTMF dialplan application
ASTERISK-15957: [patch] Updates to Application Documentation
ASTERISK-15958: [patch] [regression] Using Local channels with queues causes deadlocks
ASTERISK-15959: [patch] OOH323 Outgoing Call Fails if Originated from a DAHDI Extension
ASTERISK-15960: crash when calling ao2_unlock inside pthread_timer_disable_continuous - NOT FIXED PLEASE RE-OPEN
ASTERISK-15961: lack of locking in dahdi_request()
ASTERISK-15962: [patch] Update the doc/backtrace.txt documentation
ASTERISK-15963: Conference will not record with DAHDI loaded, but no hardware
ASTERISK-15964: [patch] Queue announcement playing times are delayed by retry+timeout seconds in slot 0 (1.4.26.2)
ASTERISK-15965: [patch] Phone keeps ringing when hangup between 'NOTIFY' and 'Status: 180 Ringing'
ASTERISK-15966: ast->tech_pvt->rtp contains garbage yielding SEGFAULT
ASTERISK-15967: [patch] Suggestion to add Seconds on both cases of Action Status of Manager
ASTERISK-15968: call distribiution problem in fewestcalls and lastrecent strategy
ASTERISK-15969: Unable to exit Directory() application with large number of users (same last name)
ASTERISK-15970: Changing storm-prevention behaviour in logger.conf
ASTERISK-15971: Core dumped
ASTERISK-15972: RecordFile API does not return after timeout
ASTERISK-15973: Asterisk core dumps using MOH
ASTERISK-15974: Over Threshold
ASTERISK-15975: [patch] Dial()'s do_forward() breaks Local/ channel frame forwarding
ASTERISK-15976: module app_voicemail not load
ASTERISK-15977: Using SORT strings with comma in them are truncated up to the first comma.
ASTERISK-15978: MOH+Madplay when going from one to another the second stream is not started
ASTERISK-15979: Crash on Hangup during post_cdr
ASTERISK-15980: Crash on Park execute
ASTERISK-15981: Asterisk consumes 100% CPU, high interupt load, calls stay at ringing state
ASTERISK-15982: Segmentation fault: blindxfer (#) using SIP
ASTERISK-15983: Segmentation fault and restart asterisk
ASTERISK-15984: Unabled to xfer call picked up from ParkedCall
ASTERISK-15985: Exceptionally long voice queuing when using chan\Local to playback Extensions
ASTERISK-15986: [patch] Segfault on ael parsing
ASTERISK-15987: [patch] Deadlock between ast_hangup and pri_dchannel
ASTERISK-15988: 100% CPU load on debian when loading from initscript
ASTERISK-15989: All channels get Congestion following PTP MDL can't handle error of type F message
ASTERISK-15990: [patch] Missing Menuselect Option "Compiler Flags - Development" in dev mode
ASTERISK-15991: [patch] Add ability to generate an ASCII document from the TeX files
ASTERISK-15992: [patch] Originate Action output is inconsistent with other manager actions
ASTERISK-15993: [patch] Colourized logging option request as an option
ASTERISK-15994: [patch] System() taking excessive time to return with nonroot
ASTERISK-15995: Asterisk crashes sometimes (device state?)
ASTERISK-15996: [patch] using opal instead of pwlib/openh323
ASTERISK-15997: [patch] Segmentation fault with unanswered inbound call via chan_ooh323
ASTERISK-15998: Asterisk 1.4.29 crashes in astobj2.c
ASTERISK-15999: contexts and voicemail